Glossary

WebRTC (Web Real-Time Communication)

With WebRTC, you can add real-time communication capabilities to your application that works on top of an open standard. It supports video, voice, and generic data to be sent between peers, allowing developers to build powerful voice and video communication solutions. The technology is available on all modern browsers as well as on native clients for all major platforms. The technologies behind WebRTC are implemented as an open web standard and available as regular JavaScript APIs in all major browsers. For native clients, like Android and iOS applications, a library is available that provides the same functionality. The WebRTC project is open-source and supported by Apple, Google, Microsoft and Mozilla, amongst others.

There are many different use-cases for WebRTC, from basic web apps that uses the camera or microphone, to more advanced video-calling applications and screen sharing. We have gathered a number of code samples to better illustrate how the technology works and what you can use it for.

WebRTC API on mozilla.org


Melrose Labs Voice Gateway/Conference
Melrose Labs Video Conferencing

Melrose Labs: 我们的使命是为企业,短信聚合器和消息传递提供商提供跨移动,固定电话和互联网的云通信中的关键服务。这些服务使组织能够在向客户提供服务,可靠性,对服务的洞察力以及提高的运营效率方面取得飞跃。